Displaying 20 results from an estimated 4000 matches similar to: "Registrations stops after 403 FORBIDDEN"
2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER
2009 Oct 25
2
help sip show on CLI : no such command
What is wrong when I can not execute any command that starts with
sip ???
> freepbx*CLI> help sip show
> No such command 'sip show'.
> freepbx*CLI> help sip
> No such command 'sip'.
IAX works fine :
> freepbx*CLI> help iax
> iax2 provision Provision an IAX device
> iax2 prune realtime Prune a cached realtime lookup
>
2016 Aug 17
4
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 17:45, George Joseph wrote:
>
>
> On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> On 16-08-16 04:38, George Joseph wrote:
>>
>>
>> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
>> <jonas.kellens at telenet.be <mailto:jonas.kellens at
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2016 Aug 16
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 04:38, George Joseph wrote:
>
>
> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> using pjproject 2.5.5
> using asterisk-certified-13.8-cert1
>
>
> IIRC there were API changes in pjproject 2.5 that aren't accounted for
> in
2016 Aug 11
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 11-08-16 18:03, Matt Fredrickson wrote:
> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote:
>> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
>> functionality as there are certain functions deprecated/replaced. This can
>> also cause headache :-)
>>
>> I will do so if there is no other option.
2010 Oct 26
11
Auto provisioning from public server
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address in the
provisioning link to obtain SIP username / password settings ?
Kind regards,
Jonas.
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2016 Aug 12
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-16 09:25, ????? ?????? wrote:
>
> Try delete nat from 770000wrtc settings ice should do the same
>
>
> On Aug 11, 2016 10:00 PM, "Jonas
2010 Jun 24
3
Very strange registration problem
Hello list,
using asterisk 1.4.30
I have the strangest problem that some SIP accounts can register to my
Asterisk and others not. I see no connection between all those that can
register or all those that can't.
It's not a firewall problem as all register to port 5060 and the range
5060 --> 5064 is open.
It's just very strange that some can register and other not.
Any
2016 Sep 10
2
Queue show : failed to extend from 240 to 327
On 10-09-16 00:50, Richard Mudgett wrote:
>
>
> On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> when I type on the Asterisk CLi 'queue show', I first get a list
> of my queues and then the following :
>
>
> failed to extend from 240 to 327
2016 Nov 21
3
Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 15:17, Matthew Jordan wrote:
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> when using Asterisk version 13.12.2 I notice that it takes up to
> 30 seconds (sometimes even longer) for a call queue to call its
> members.
>
>
2012 Feb 02
1
MixMonitor and ChanSpy
Hello,
ChanSpy can not be used on a Channel that is being recorded with
MixMonitor.
How can I verify if a channel which I want to spy on, is currently not
being recorded ?!
Kind regards,
Jonas.
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2016 Aug 11
3
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This
can also cause headache :-)
I will do so if there is no other option.
But still, I don't see why Ast 13 would differ so much in this case ? If
ICE and NAT is working (not causing problems) why should Ast 13 bring me
audio and Ast 12 don't
2011 Mar 24
1
Fwd: Asterisk 1.6.2.10 & CDR custom added field
Hello,
is there anyone who can point me to correct information ?
Following http://pbxinaflash.com/forum/showthread.php?t=9042 and
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR
does not result in a working environment for me.
Any feedback appreciated.
Kind regards,
Jonas.
-------- Original Message --------
Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR
2011 May 03
1
audiohook.c: Failed to get 160 samples from write factory
Hello,
I see a lot of these messages in the debug log :
/[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples
from write factory 0xae17e18
[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples
from write factory 0xae17e18
[May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and
write factory 0xae17e18 both fail to provide 160 samples
[May 3
2012 Sep 28
1
Disconnect calls : known reasons
Hello,
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call ending, as if one of the connected parties hung up
but that is not the case !
So what could be a bottleneck ? Any known reasons for random hangup ?
2011 Apr 02
1
Registration from '"000000" x 1000
Hello list,
I often see the following in my message log :
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"000000"
<sip:000000 at MY-IP>' failed for '184.106.109.168' - No matching peer found
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"000000"
<sip:000000 at MY-IP>' failed for '184.106.109.168' -
2012 Mar 07
1
Finish ChanSpy() when channel spied hangs up
Is there any way to do this?
Thanks
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2012 May 08
4
Asterisk 1.8 Transfer CallerID
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B sees the number of colleague A on his
screen while talking to the external calling number.
I expect here that colleague B would see the external calling number on
the screen